1. Implementation Lessons using WebRTC in Asterisk
Astricon, October 2013
Moisés Silva
Manager, Software Engineering

2. Agenda WebRTC Intro WebRTC Asterisk Architecture Install & Config
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WebRTC Intro
WebRTC Asterisk Architecture
Install & Config
Troubleshooting

3. WebRTC Intro It is not a phone in the browser!
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It is not a phone in the browser!

4. WebRTC Intro It is a full RTC engine in the browser!
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It is a full RTC engine in the browser!

5. WebRTC Intro Yes, it can be used for a phone in the browser 
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Yes, it can be used for a phone in the browser 

6. WebRTC Intro Full media engine API in the web browser
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Full media engine API in the web browser
No “call” or “session” signaling defined
Generic data interchange between browsers, peer to peer
State of the art NAT traversal techniques

7. WebRTC Intro WebRTC comes with multiple APIs, ie:
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WebRTC comes with multiple APIs, ie:
Peer-to-Peer Connections (RTCPeerConnection)
Peer-to-Peer Data API (RTCDataChannel)
Statistics (RTCStats)
Media Stream (getUserMedia)

8. WebRTC Intro WebRTC uses established protocols:
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WebRTC uses established protocols:
SRTP/SRTCP for media exchange (secure RTP)
SDP (its use is controversial and currently challenged)
ICE, STUN, TURN for NAT Traversal
DTLS for key exchange
G.711, Opus, VP8/H.264 etc; for voice and video

9. WebRTC Intro What signaling to use is up to you: SIP XMPP/Jingle
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What signaling to use is up to you:
SIP
XMPP/Jingle
RESTful API (json)
OpenPeer ….

10. WebRTC Intro Applications A phone, video calls, conferencing etc!
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Applications
A phone, video calls, conferencing etc!
Video games
P2P Video Streaming (Chromecast)
Motion-detecting Baby Monitor (

11. WebRTC Intro WebRTC Web Triangle Web Signaling Signaling
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WebRTC Web Triangle
Web
Signaling
Signaling
Encrypted Media
Alice’s Browser
Bob’s Browser

12. WebRTC in Asterisk SIP over WS SIP over WS Encrypted Media
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SIP over WS
SIP over WS
Encrypted Media
Encrypted Media
Alice’s Browser
Bob’s Browser

13. WebRTC in Asterisk WebRTC Gateway SIP/RTP, Jingle, FXO/FXS,
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WebRTC Gateway
SIP/RTP, Jingle, FXO/FXS,
PRI, SS7 etc …
SIP over WS
Encrypted Media
Alice’s Browser

14. WebRTC in Asterisk Javascript SIP res_http_websocket chan_sip WebRTC
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Javascript SIP
res_http_websocket
chan_sip
WebRTC
res_rtp_asterisk
res_srtp

15. WebRTC in Asterisk sipml5 Asterisk 11 Chrome 30
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sipml5
Asterisk 11
Chrome 30

16. Installing WebRTC Support
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Make sure you have:
libuuid-devel (required by res_rtp_asterisk)
OpenSSL w/ DTLS support (1.0.1e has SSL_CTX_set_tlsext_use_srtp)
libsrtp-devel

17. Installing WebRTC Support
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Easy usual steps …
./configure
make menuselect:
res_http_websocket
res_rtp_asterisk
make install

18. Configuring WebRTC Support
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Enable the websockets server (http.conf)
enabled=yes
bindaddr=
bindport=8088

19. Configuring WebRTC Support
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Good idea to use secure websockets (http.conf)
tlsenable=yes
tlsbindaddr= :8089
tlscertfile=localhost.crt
tlsprivatekey=localhost.key

20. Configuring WebRTC Support
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But … Asterisk currently seems to have issues with secure WebSockets, patches available to fix them

21. Configuring WebRTC Support
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Verify the HTTP server status

22. Configuring WebRTC Support
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Test websockets connectivity
npm install –g ws
wscat –s echo –c ws://<host>:<port>/ws wscat –s echo –c wss://<host>:<port>/ws

23. Configuring WebRTC Support
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Test websockets connectivity

24. Configuring WebRTC Support
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25. Configuring WebRTC Support
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Enable SIP over websockets (sip.conf)
transport=udp,ws,wss
Make sure you use the /ws URL when connecting from JavaScript

26. Configuring WebRTC Support
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Testing using sipml5.org/call.htm

27. Configuring WebRTC Support
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28. Troubleshooting Troubleshooting Toolkit javascript console
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Troubleshooting Toolkit
javascript console
chrome://webrtc-internals
Node ws (test websockets)
Wireshark!

29. Troubleshooting The javascript console is your friend
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The javascript console is your friend

30. Troubleshooting Checking out chrome://webrtc-internals
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Checking out chrome://webrtc-internals

31. Troubleshooting Checking out chrome://webrtc-internals
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Checking out chrome://webrtc-internals

32. Troubleshooting
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Note that Wireshark VoIP calls menu won’t work for calls over websockets
You can however use “Follow TCP stream” and see the entire SIP signaling flow
RTP decoding will not work either (rtcp-mux)

33. Troubleshooting
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TLS decryption can be achieved by installing the private key on Wireshark
Preferences -> Protocols -> SSL -> RSA Key List

34. Troubleshooting Wireshark decrypted secure WebSocket payload
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Wireshark decrypted secure WebSocket payload

35. Future Trickle Ice Use of other codecs (ie Opus, iSAC) Video (VP8)
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Trickle Ice
Use of other codecs (ie Opus, iSAC)
Video (VP8)
Use libwebsockets in res_http_websocket?

36. Conclusion Asterisk + WebRTC gateway is easy to setup!
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Asterisk + WebRTC gateway is easy to setup!
Know your debugging tools
Understand the protocols involved
Have fun and hack away!

37. QUESTIONS

38. Contact Us Sangoma Technologies Website http://www.sangoma.com/
100 Renfrew Drive, Suite 100 Markham, Ontario L3R 9R6 Canada
Website
Telephone
x2 (for Sales)

39. THANK YOU